[alsa-devel] [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation
Lokesh Vutla
lokeshvutla at ti.com
Mon Mar 10 10:39:40 CET 2014
Hi Jyri,
Just a few minor comments...
On Monday 10 March 2014 02:22 PM, Jyri Sarha wrote:
> This commit adds a bare bones driver support for TLV320AIC31XX family
> audio codecs. The driver adds basic stereo playback trough headphone
> and speaker outputs and mono capture trough microphone inputs.
>
> The driver is currently missing support at least for mini DSP features
> and jack detection. I have tested the driver only on TLV320AIC3111,
> but based on the data sheets TLV320AIC3100, TLV320AIC3110, and
> TLV320AIC3120 should work Ok too.
>
> The base for the implementation was taken from:
> git at gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx
> -branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13.
>
> Signed-off-by: Jyri Sarha <jsarha at ti.com>
> ---
> .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 +
> include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 +
> sound/soc/codecs/Kconfig | 4 +
> sound/soc/codecs/Makefile | 2 +
> sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++
> sound/soc/codecs/tlv320aic31xx.h | 258 ++++
> 6 files changed, 1677 insertions(+)
> create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
> create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h
> create mode 100644 sound/soc/codecs/tlv320aic31xx.c
> create mode 100644 sound/soc/codecs/tlv320aic31xx.h
>
> diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
> new file mode 100644
> index 0000000..0109df1
> --- /dev/null
> +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
> @@ -0,0 +1,61 @@
> +Texas Instruments - tlv320aic31xx Codec module
> +
> +The tlv320aic31xx serial control bus communicates through I2C protocols
> +
> +Required properties:
> +
> +- compatible - "string" - One of:
> + "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp
> + "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp
> + "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP)
> + "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
> + "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
> + "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
> +
> +- reg - <int> - I2C slave address
> +
> +
> +Optional properties:
> +
> +- gpio-reset - gpio pin number used for codec reset
> +- ai31xx-micbias-vg - MicBias Voltage setting
> + 0 or MICBIAS_OFF - MICBIAS output it not powered
> + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V
> + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V
> + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD
> + If this node is not mentioned or if the value is unknown, then
> + micbias is set to 2.0V.
> +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
> + DVDD-supply : power supplies for the device as covered in
> + Documentation/devicetree/bindings/regulator/regulator.txt
> +
> +CODEC output pins:
> + * HPL
> + * HPR
> + * SPL, devices with stereo speaker amp
> + * SPR, devices with stereo speaker amp
> + * SPK, devices with mono speaker amp
> +
> +CODEC input pins:
> + * MIC1LP
> + * MIC1RP
> + * MIC1LM
> +
> +The pins can be used in referring sound node's audio-routing property.
> +
> +Example:
> +#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
> +
> +tlv320aic31xx: tlv320aic31xx at 18 {
> + compatible = "ti,tlv320aic311x";
> + reg = <0x18>;
> +
> + ai31xx-micbias-vg = <MICBIAS_OFF>;
> +
> + HPVDD-supply = <®ulator>;
> + SPRVDD-supply = <®ulator>;
> + SPLVDD-supply = <®ulator>;
> + AVDD-supply = <®ulator>;
> + IOVDD-supply = <®ulator>;
> + DVDD-supply = <®ulator>;
> +};
> diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h
> new file mode 100644
> index 0000000..da9360f
> --- /dev/null
> +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h
> @@ -0,0 +1,9 @@
> +#ifndef __DT_TLV320AIC31XX_MICBIAS_H
> +#define __DT_TLV320AIC31XX_MICBIAS_H
> +
> +#define MICBIAS_OFF 0
> +#define MICBIAS_2_0V 1
> +#define MICBIAS_2_5V 2
> +#define MICBIAS_AVDDV 3
> +
> +#endif /* __DT_TLV320AIC31XX_MICBIAS_H */
> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
> index e19b64f..af3c049 100644
> --- a/sound/soc/codecs/Kconfig
> +++ b/sound/soc/codecs/Kconfig
> @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS
> select SND_SOC_TAS5086 if I2C
> select SND_SOC_TLV320AIC23 if I2C
> select SND_SOC_TLV320AIC26 if SPI_MASTER
> + select SND_SOC_TLV320AIC31XX if I2C
> select SND_SOC_TLV320AIC32X4 if I2C
> select SND_SOC_TLV320AIC3X if I2C
> select SND_SOC_TPA6130A2 if I2C
> @@ -428,6 +429,9 @@ config SND_SOC_TLV320AIC26
> tristate
> depends on SPI
>
> +config SND_SOC_TLV320AIC31XX
> + tristate
> +
> config SND_SOC_TLV320AIC32X4
> tristate
>
> diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
> index 90ab24b..018e3f3 100644
> --- a/sound/soc/codecs/Makefile
> +++ b/sound/soc/codecs/Makefile
> @@ -75,6 +75,7 @@ snd-soc-stac9766-objs := stac9766.o
> snd-soc-tas5086-objs := tas5086.o
> snd-soc-tlv320aic23-objs := tlv320aic23.o
> snd-soc-tlv320aic26-objs := tlv320aic26.o
> +snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
> snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
> snd-soc-tlv320aic3x-objs := tlv320aic3x.o
> snd-soc-tlv320dac33-objs := tlv320dac33.o
> @@ -215,6 +216,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
> obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o
> obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
> obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
> +obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o
> obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o
> obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
> obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
> diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
> new file mode 100644
> index 0000000..1d3d2416
> --- /dev/null
> +++ b/sound/soc/codecs/tlv320aic31xx.c
> @@ -0,0 +1,1343 @@
> +/*
> + * ALSA SoC TLV320AIC31XX codec driver
> + *
> + * Copyright (C) 2013 Texas Instruments, Inc.
2014..:)
> + *
> + * Author: Jyri Sarha <jsarha at ti.com>
> + *
> + * Based on ground work by: Ajit Kulkarni <x0175765 at ti.com>
> + *
> + * This package is free software; you can redistribute it and/or modify
> + * it under the terms of the GNU General Public License version 2 as
> + * published by the Free Software Foundation.
> + *
> + * THIS PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR
> + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
> + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
> + *
> + * The TLV320AIC31xx series of audio codec is a low-power, highly integrated
> + * high performance codec which provides a stereo DAC, a mono ADC,
> + * and mono/stereo Class-D speaker driver.
> + */
> +
> +#include <linux/module.h>
> +#include <linux/moduleparam.h>
> +#include <linux/init.h>
> +#include <linux/delay.h>
> +#include <linux/pm.h>
> +#include <linux/i2c.h>
> +#include <linux/gpio.h>
> +#include <linux/regulator/consumer.h>
> +#include <linux/of_gpio.h>
> +#include <linux/slab.h>
> +#include <sound/core.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/soc.h>
> +#include <sound/initval.h>
> +#include <sound/tlv.h>
> +#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
> +
> +#include "tlv320aic31xx.h"
> +
> +static const struct reg_default aic31xx_reg_defaults[] = {
> + { AIC31XX_CLKMUX, 0x00 },
> + { AIC31XX_PLLPR, 0x11 },
> + { AIC31XX_PLLJ, 0x04 },
> + { AIC31XX_PLLDMSB, 0x00 },
> + { AIC31XX_PLLDLSB, 0x00 },
> + { AIC31XX_NDAC, 0x01 },
> + { AIC31XX_MDAC, 0x01 },
> + { AIC31XX_DOSRMSB, 0x00 },
> + { AIC31XX_DOSRLSB, 0x80 },
> + { AIC31XX_NADC, 0x01 },
> + { AIC31XX_MADC, 0x01 },
> + { AIC31XX_AOSR, 0x80 },
> + { AIC31XX_IFACE1, 0x00 },
> + { AIC31XX_DATA_OFFSET, 0x00 },
> + { AIC31XX_IFACE2, 0x00 },
> + { AIC31XX_BCLKN, 0x01 },
> + { AIC31XX_DACSETUP, 0x14 },
> + { AIC31XX_DACMUTE, 0x0c },
> + { AIC31XX_LDACVOL, 0x00 },
> + { AIC31XX_RDACVOL, 0x00 },
> + { AIC31XX_ADCSETUP, 0x00 },
> + { AIC31XX_ADCFGA, 0x80 },
> + { AIC31XX_ADCVOL, 0x00 },
> + { AIC31XX_HPDRIVER, 0x04 },
> + { AIC31XX_SPKAMP, 0x06 },
> + { AIC31XX_DACMIXERROUTE, 0x00 },
> + { AIC31XX_LANALOGHPL, 0x7f },
> + { AIC31XX_RANALOGHPR, 0x7f },
> + { AIC31XX_LANALOGSPL, 0x7f },
> + { AIC31XX_RANALOGSPR, 0x7f },
> + { AIC31XX_HPLGAIN, 0x02 },
> + { AIC31XX_HPRGAIN, 0x02 },
> + { AIC31XX_SPLGAIN, 0x00 },
> + { AIC31XX_SPRGAIN, 0x00 },
> + { AIC31XX_MICBIAS, 0x00 },
> + { AIC31XX_MICPGA, 0x80 },
> + { AIC31XX_MICPGAPI, 0x00 },
> + { AIC31XX_MICPGAMI, 0x00 },
> +};
> +
> +static bool aic31xx_volatile(struct device *dev, unsigned int reg)
> +{
> + switch (reg) {
> + case AIC31XX_PAGECTL: /* regmap implementation requires this */
> + case AIC31XX_RESET: /* always clears after write */
> + case AIC31XX_OT_FLAG:
> + case AIC31XX_ADCFLAG:
> + case AIC31XX_DACFLAG1:
> + case AIC31XX_DACFLAG2:
> + case AIC31XX_OFFLAG: /* Sticky interrupt flags */
> + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */
> + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */
> + case AIC31XX_INTRDACFLAG2:
> + case AIC31XX_INTRADCFLAG2:
> + return true;
> + }
> + return false;
> +}
> +
> +static bool aic31xx_writeable(struct device *dev, unsigned int reg)
> +{
> + switch (reg) {
> + case AIC31XX_OT_FLAG:
> + case AIC31XX_ADCFLAG:
> + case AIC31XX_DACFLAG1:
> + case AIC31XX_DACFLAG2:
> + case AIC31XX_OFFLAG: /* Sticky interrupt flags */
> + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */
> + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */
> + case AIC31XX_INTRDACFLAG2:
> + case AIC31XX_INTRADCFLAG2:
> + return false;
> + }
> + return true;
> +}
> +
> +static const struct regmap_range_cfg aic31xx_ranges[] = {
> + {
> + .range_min = 0,
> + .range_max = 12 * 128,
> + .selector_reg = AIC31XX_PAGECTL,
> + .selector_mask = 0xff,
> + .selector_shift = 0,
> + .window_start = 0,
> + .window_len = 128,
> + },
> +};
> +
> +struct regmap_config aic31xx_i2c_regmap = {
> + .reg_bits = 8,
> + .val_bits = 8,
> + .writeable_reg = aic31xx_writeable,
> + .volatile_reg = aic31xx_volatile,
> + .reg_defaults = aic31xx_reg_defaults,
> + .num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults),
> + .cache_type = REGCACHE_RBTREE,
> + .ranges = aic31xx_ranges,
> + .num_ranges = ARRAY_SIZE(aic31xx_ranges),
> + .max_register = 12 * 128,
> +};
> +
> +#define AIC31XX_NUM_SUPPLIES 6
> +static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = {
> + "HPVDD",
> + "SPRVDD",
> + "SPLVDD",
> + "AVDD",
> + "IOVDD",
> + "DVDD",
> +};
> +
> +struct aic31xx_disable_nb {
> + struct notifier_block nb;
> + struct aic31xx_priv *aic31xx;
> +};
> +
> +struct aic31xx_priv {
> + struct snd_soc_codec *codec;
> + u8 i2c_regs_status;
> + struct device *dev;
> + struct regmap *regmap;
> + struct aic31xx_pdata pdata;
> + struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES];
> + struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES];
> + unsigned int sysclk;
> + int rate_div_line;
> + int codec_clk_on;
> +};
> +
> +struct aic31xx_rate_divs {
> + u32 mclk;
> + u32 rate;
> + u8 p_val;
> + u8 pll_j;
> + u16 pll_d;
> + u16 dosr;
> + u8 ndac;
> + u8 mdac;
> + u8 aosr;
> + u8 nadc;
> + u8 madc;
> +};
> +
> +/* ADC dividers can be disabled by cofiguring them to 0 */
> +static const struct aic31xx_rate_divs aic31xx_divs[] = {
> + /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */
> + /* 8k rate */
> + {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2},
> + {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2},
> + {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2},
> + /* 11.025k rate */
> + {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2},
> + {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2},
> + {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2},
> + /* 16k rate */
> + {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2},
> + {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2},
> + {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2},
> + /* 22.05k rate */
> + {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2},
> + {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2},
> + {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2},
> + /* 32k rate */
> + {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2},
> + {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2},
> + {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2},
> + /* 44.1k rate */
> + {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2},
> + {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2},
> + {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2},
> + /* 48k rate */
> + {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2},
> + {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2},
> + {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2},
> + /* 88.2k rate */
> + {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2},
> + {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2},
> + {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2},
> + /* 96k rate */
> + {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2},
> + {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2},
> + {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2},
> + /* 176.4k rate */
> + {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2},
> + {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2},
> + {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2},
> + /* 192k rate */
> + {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2},
> + {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2},
> + {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2},
> +};
> +
> +static const char * const ldac_in_text[] = {
> + "Off", "Left Data", "Right Data", "Mono"
> +};
> +
> +static const char * const rdac_in_text[] = {
> + "Off", "Right Data", "Left Data", "Mono"
> +};
> +
> +static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text);
> +
> +static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text);
> +
> +static const char * const mic_select_text[] = {
> + "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm"
> +};
> +
> +static const
> +SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text);
> +static const
> +SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text);
> +static const
> +SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text);
> +
> +static const
> +SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
> +static const
> +SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text);
> +
> +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0);
> +static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0);
> +static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0);
> +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0);
> +static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0);
> +static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0);
> +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0);
> +static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0);
> +
> +/*
> + * controls to be exported to the user space
> + */
> +static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
> + SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL,
> + AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv),
> +
> + SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1,
> + adc_fgain_tlv),
> +
> + SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1),
> + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL,
> + 0, -24, 40, 6, 0, adc_cgain_tlv),
> +
> + SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0,
> + 119, 0, mic_pga_tlv),
> +
> + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
> + AIC31XX_HPRGAIN, 2, 1, 0),
> + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
> + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),
> +
> + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
> + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
> +};
> +
> +static const struct snd_kcontrol_new aic311x_snd_controls[] = {
> + SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN,
> + AIC31XX_SPRGAIN, 2, 1, 0),
> + SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN,
> + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv),
> +
> + SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL,
> + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv),
> +};
> +
> +static const struct snd_kcontrol_new aic310x_snd_controls[] = {
> + SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN,
> + 2, 1, 0),
> + SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN,
> + 3, 3, 0, class_D_drv_tlv),
> +
> + SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL,
> + 0, 0x7F, 1, sp_vol_tlv),
> +};
> +
> +static const struct snd_kcontrol_new ldac_in_control =
> + SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum);
> +
> +static const struct snd_kcontrol_new rdac_in_control =
> + SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum);
> +
> +int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg,
> + unsigned int mask, unsigned int wbits, int sleep,
> + int count)
> +{
> + unsigned int bits;
> + int counter = count;
> + int ret = regmap_read(aic31xx->regmap, reg, &bits);
> + while ((bits & mask) != wbits && counter && !ret) {
> + usleep_range(sleep, sleep * 2);
> + ret = regmap_read(aic31xx->regmap, reg, &bits);
> + counter--;
> + }
> + if ((bits & mask) != wbits) {
> + dev_err(aic31xx->dev,
> + "%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n",
> + __func__, reg, bits, wbits, ret, mask,
> + (count - counter) * sleep);
> + ret = -1;
> + }
> + return ret;
> +}
> +
> +
extra line...
> +#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg))
> +
> +static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
> + struct snd_kcontrol *kcontrol, int event)
> +{
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec);
> + unsigned int reg = AIC31XX_DACFLAG1;
> + unsigned int mask;
> +
> + dev_dbg(w->codec->dev, "%s -> %s (event: %d)\n", w->name,
> + event == SND_SOC_DAPM_POST_PMU ? "on" :
> + event == SND_SOC_DAPM_POST_PMD ? "off" :
> + "(unsupported event)", event);
> +
> + switch (WIDGET_BIT(w->reg, w->shift)) {
> + case WIDGET_BIT(AIC31XX_DACSETUP, 7):
> + mask = AIC31XX_LDACPWRSTATUS_MASK;
> + break;
> + case WIDGET_BIT(AIC31XX_DACSETUP, 6):
> + mask = AIC31XX_RDACPWRSTATUS_MASK;
> + break;
> + case WIDGET_BIT(AIC31XX_HPDRIVER, 7):
> + mask = AIC31XX_HPLDRVPWRSTATUS_MASK;
> + break;
> + case WIDGET_BIT(AIC31XX_HPDRIVER, 6):
> + mask = AIC31XX_HPRDRVPWRSTATUS_MASK;
> + break;
> + case WIDGET_BIT(AIC31XX_SPKAMP, 7):
> + mask = AIC31XX_SPLDRVPWRSTATUS_MASK;
> + break;
> + case WIDGET_BIT(AIC31XX_SPKAMP, 6):
> + mask = AIC31XX_SPRDRVPWRSTATUS_MASK;
> + break;
> + case WIDGET_BIT(AIC31XX_ADCSETUP, 7):
> + mask = AIC31XX_ADCPWRSTATUS_MASK;
> + reg = AIC31XX_ADCFLAG;
> + break;
> + default:
> + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",
> + w->name, __func__);
> + return -EINVAL;
> + }
> +
> + switch (event) {
> + case SND_SOC_DAPM_POST_PMU:
> + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100);
> + case SND_SOC_DAPM_POST_PMD:
> + return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100);
> + }
> +
> + dev_dbg(w->codec->dev, "Unhandled dapm widget event %d from %s\n",
> + event, w->name);
> + return 0;
> +}
> +
> +static const struct snd_kcontrol_new left_output_switches[] = {
> + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
> + SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0),
> + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0),
> +};
> +
> +static const struct snd_kcontrol_new right_output_switches[] = {
> + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
> + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0),
> +};
> +
> +static const struct snd_kcontrol_new p_term_mic1lp =
> + SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum);
> +
> +static const struct snd_kcontrol_new p_term_mic1rp =
> + SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum);
> +
> +static const struct snd_kcontrol_new p_term_mic1lm =
> + SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum);
> +
> +static const struct snd_kcontrol_new m_term_mic1lm =
> + SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum);
> +
> +static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch =
> + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0);
> +
> +static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch =
> + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0);
> +
> +static const struct snd_kcontrol_new aic31xx_dapm_spl_switch =
> + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0);
> +
> +static const struct snd_kcontrol_new aic31xx_dapm_spr_switch =
> + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0);
> +
> +static int mic_bias_event(struct snd_soc_dapm_widget *w,
> + struct snd_kcontrol *kcontrol, int event)
> +{
> + struct snd_soc_codec *codec = w->codec;
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + switch (event) {
> + case SND_SOC_DAPM_POST_PMU:
> + /* change mic bias voltage to user defined */
> + if (aic31xx->pdata.micbias_vg != MICBIAS_OFF) {
> + snd_soc_update_bits(codec, AIC31XX_MICBIAS,
> + AIC31XX_MICBIAS_MASK,
> + aic31xx->pdata.micbias_vg <<
> + AIC31XX_MICBIAS_SHIFT);
> + dev_dbg(codec->dev, "%s: turned on\n", __func__);
> + }
> + break;
> + case SND_SOC_DAPM_PRE_PMD:
> + if (aic31xx->pdata.micbias_vg != MICBIAS_OFF) {
> + snd_soc_update_bits(codec, AIC31XX_MICBIAS,
> + AIC31XX_MICBIAS_MASK, MICBIAS_OFF);
> + dev_dbg(codec->dev, "%s: turned off\n", __func__);
> + }
> + break;
> + }
> + return 0;
> +}
> +
> +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
> + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
> +
> + SND_SOC_DAPM_MUX("DAC Left Input",
> + SND_SOC_NOPM, 0, 0, &ldac_in_control),
> + SND_SOC_DAPM_MUX("DAC Right Input",
> + SND_SOC_NOPM, 0, 0, &rdac_in_control),
> + /* DACs */
> + SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback",
> + AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event,
> + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
> +
> + SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback",
> + AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
> + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
> +
> + /* Output Mixers */
> + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
> + left_output_switches,
> + ARRAY_SIZE(left_output_switches)),
> + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
> + right_output_switches,
> + ARRAY_SIZE(right_output_switches)),
> +
> + SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0,
> + &aic31xx_dapm_hpl_switch),
> + SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0,
> + &aic31xx_dapm_hpr_switch),
> +
> + /* Output drivers */
> + SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0,
> + NULL, 0, aic31xx_dapm_power_event,
> + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
> + SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0,
> + NULL, 0, aic31xx_dapm_power_event,
> + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
> +
> + /* ADC */
> + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
> + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
> + SND_SOC_DAPM_POST_PMD),
> +
> + /* Input Selection to MIC_PGA */
> + SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0,
> + &p_term_mic1lp),
> + SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0,
> + &p_term_mic1rp),
> + SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0,
> + &p_term_mic1lm),
> +
> + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0,
> + &m_term_mic1lm),
> + /* Enabling & Disabling MIC Gain Ctl */
> + SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA,
> + 7, 1, NULL, 0),
> +
> + /* Mic Bias */
> + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
> + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
> +
> + /* Outputs */
> + SND_SOC_DAPM_OUTPUT("HPL"),
> + SND_SOC_DAPM_OUTPUT("HPR"),
> +
> + SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
> +
> + /* Inputs */
> + SND_SOC_DAPM_INPUT("MIC1LP"),
> + SND_SOC_DAPM_INPUT("MIC1RP"),
> + SND_SOC_DAPM_INPUT("MIC1LM"),
> +
> + SND_SOC_DAPM_INPUT("Internal ADC Source"),
> +};
> +
> +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = {
> + /* AIC3111 and AIC3110 have stereo class-D amplifier */
> + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0,
> + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
> + SND_SOC_DAPM_POST_PMD),
> + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0,
> + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
> + SND_SOC_DAPM_POST_PMD),
> + SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0,
> + &aic31xx_dapm_spl_switch),
> + SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0,
> + &aic31xx_dapm_spr_switch),
> + SND_SOC_DAPM_OUTPUT("SPL"),
> + SND_SOC_DAPM_OUTPUT("SPR"),
> +};
> +
> +/* AIC3100 and AIC3120 have only mono class-D amplifier */
> +static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = {
> + SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0,
> + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
> + SND_SOC_DAPM_POST_PMD),
> + SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0,
> + &aic31xx_dapm_spl_switch),
> + SND_SOC_DAPM_OUTPUT("SPK"),
> +};
> +
> +static const struct snd_soc_dapm_route
> +aic31xx_audio_map[] = {
> + /* Make DACs turn on when playing even if not mixed into any outputs */
> + {"Internal DAC Sink", NULL, "DAC Left"},
> + {"Internal DAC Sink", NULL, "DAC Right"},
> +
> + /* Make ADC turn on when recording even if not mixed from any inputs */
> + {"ADC", NULL, "Internal ADC Source"},
> +
> + /* DAC Input Routing */
> + {"DAC Left Input", "Left Data", "DAC IN"},
> + {"DAC Left Input", "Right Data", "DAC IN"},
> + {"DAC Left Input", "Mono", "DAC IN"},
> + {"DAC Right Input", "Left Data", "DAC IN"},
> + {"DAC Right Input", "Right Data", "DAC IN"},
> + {"DAC Right Input", "Mono", "DAC IN"},
> + {"DAC Left", NULL, "DAC Left Input"},
> + {"DAC Right", NULL, "DAC Right Input"},
> +
> + /* Mic input */
> + {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"},
> + {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"},
> + {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"},
> + {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"},
> + {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"},
> + {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"},
> + {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"},
> + {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"},
> + {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"},
> +
> + {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"},
> + {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"},
> + {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"},
> +
> + {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"},
> + {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"},
> + {"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"},
> + {"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"},
> +
> + {"ADC", NULL, "MIC_GAIN_CTL"},
> + {"MIC_GAIN_CTL", NULL, "MICBIAS"},
> +
> + /* Left Output */
> + {"Output Left", "From Left DAC", "DAC Left"},
> + {"Output Left", "From MIC1LP", "MIC1LP"},
> + {"Output Left", "From MIC1RP", "MIC1RP"},
> +
> + /* Right Output */
> + {"Output Right", "From Right DAC", "DAC Right"},
> + {"Output Right", "From MIC1RP", "MIC1RP"},
> +
> + /* HPL path */
> + {"HP Left", "Switch", "Output Left"},
> + {"HPL Driver", NULL, "HP Left"},
> + {"HPL", NULL, "HPL Driver"},
> +
> + /* HPR path */
> + {"HP Right", "Switch", "Output Right"},
> + {"HPR Driver", NULL, "HP Right"},
> + {"HPR", NULL, "HPR Driver"},
> +};
> +
> +static const struct snd_soc_dapm_route
> +aic311x_audio_map[] = {
> + /* SP L path */
> + {"Speaker Left", "Switch", "Output Left"},
> + {"SPL ClassD", NULL, "Speaker Left"},
> + {"SPL", NULL, "SPL ClassD"},
> +
> + /* SP R path */
> + {"Speaker Right", "Switch", "Output Right"},
> + {"SPR ClassD", NULL, "Speaker Right"},
> + {"SPR", NULL, "SPR ClassD"},
> +};
> +
> +static const struct snd_soc_dapm_route
> +aic310x_audio_map[] = {
> + /* SP L path */
> + {"Speaker", "Switch", "Output Left"},
> + {"SPK ClassD", NULL, "Speaker"},
> + {"SPK", NULL, "SPK ClassD"},
> +};
> +
> +static int aic31xx_add_controls(struct snd_soc_codec *codec)
> +{
> + int err = 0;
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> +
> + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) {
> + err = snd_soc_add_codec_controls(
> + codec, aic311x_snd_controls,
> + ARRAY_SIZE(aic311x_snd_controls));
> + if (err < 0)
> + dev_dbg(codec->dev, "Invalid control\n");
> +
> + } else {
> + err = snd_soc_add_codec_controls(
> + codec, aic310x_snd_controls,
> + ARRAY_SIZE(aic310x_snd_controls));
> + if (err < 0)
> + dev_dbg(codec->dev, "Invalid Control\n");
> + }
> + return 0;
> +}
> +
> +static int aic31xx_add_widgets(struct snd_soc_codec *codec)
> +{
> + struct snd_soc_dapm_context *dapm = &codec->dapm;
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + int ret = 0;
> +
> + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) {
> + ret = snd_soc_dapm_new_controls(
> + dapm, aic311x_dapm_widgets,
> + ARRAY_SIZE(aic311x_dapm_widgets));
> + if (ret)
> + dev_err(codec->dev,
> + "Adding %d dapm widgets failed: %d\n",
> + ARRAY_SIZE(aic311x_dapm_widgets), ret);
> + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map,
> + ARRAY_SIZE(aic311x_audio_map));
> + if (ret)
> + dev_err(codec->dev,
> + "Adding %d DAPM routes failed: %d\n",
> + ARRAY_SIZE(aic311x_audio_map), ret);
> + } else {
> + ret = snd_soc_dapm_new_controls(
> + dapm, aic310x_dapm_widgets,
> + ARRAY_SIZE(aic310x_dapm_widgets));
> + if (ret)
> + dev_err(codec->dev,
> + "Adding %d dapm widgets failed: %d\n",
> + ARRAY_SIZE(aic310x_dapm_widgets), ret);
> + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map,
> + ARRAY_SIZE(aic310x_audio_map));
> + if (ret)
> + dev_err(codec->dev,
> + "Adding %d DAPM routes failed: %d\n",
> + ARRAY_SIZE(aic310x_audio_map), ret);
> + }
> +
> + return 0;
> +}
> +
> +static int aic31xx_setup_pll(struct snd_soc_codec *codec,
> + struct snd_pcm_hw_params *params)
> +{
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + int bclk_n = 0;
> + int i;
> +
> + /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */
> + snd_soc_update_bits(codec, AIC31XX_CLKMUX,
> + AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL);
> + snd_soc_update_bits(codec, AIC31XX_IFACE2,
> + AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK);
> +
> + for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) {
> + if (aic31xx_divs[i].rate == params_rate(params) &&
> + aic31xx_divs[i].mclk == aic31xx->sysclk)
> + break;
> + }
> +
> + if (i == ARRAY_SIZE(aic31xx_divs)) {
> + dev_err(codec->dev, "%s: Sampling rate %u not supported\n",
> + __func__, params_rate(params));
> + return -EINVAL;
> + }
> +
> + if (aic31xx->codec_clk_on) {
> + int old = aic31xx->rate_div_line;
> + if (aic31xx_divs[i].pll_j != aic31xx_divs[old].pll_j ||
> + aic31xx_divs[i].pll_d != aic31xx_divs[old].pll_d ||
> + aic31xx_divs[i].p_val != aic31xx_divs[old].p_val) {
> + dev_err(codec->dev,
> + "%s: Can't reconfigure PLL while it is on.\n",
> + __func__);
> + return -EBUSY;
> + }
> + } else {
> + /* PLL configuration */
> + snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK,
> + (aic31xx_divs[i].p_val << 4) | 0x01);
> + snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j);
> +
> + snd_soc_write(codec, AIC31XX_PLLDMSB,
> + aic31xx_divs[i].pll_d >> 8);
> + snd_soc_write(codec, AIC31XX_PLLDLSB,
> + aic31xx_divs[i].pll_d & 0xff);
> + }
> +
> + /* DAC dividers configuration */
> + snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK,
> + aic31xx_divs[i].ndac);
> + snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK,
> + aic31xx_divs[i].mdac);
> +
> + snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8);
> + snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff);
> +
> + /* ADC dividers configuration. Write reset value 1 if not used. */
> + snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK,
> + aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1);
> + snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK,
> + aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1);
> +
> + snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr);
> +
> + /* Bit clock divider configuration. */
> + bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac)
> + / snd_soc_params_to_frame_size(params);
> + if (bclk_n == 0) {
> + dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n",
> + __func__);
> + return -EINVAL;
> + }
> +
> + snd_soc_update_bits(codec, AIC31XX_BCLKN,
> + AIC31XX_PLL_MASK, bclk_n);
> +
> + aic31xx->rate_div_line = i;
> +
> + dev_dbg(codec->dev,
> + "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n",
> + aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d,
> + aic31xx_divs[i].p_val, aic31xx_divs[i].dosr,
> + aic31xx_divs[i].ndac, aic31xx_divs[i].mdac,
> + aic31xx_divs[i].aosr, aic31xx_divs[i].nadc,
> + aic31xx_divs[i].madc, bclk_n);
> +
> + return 0;
> +}
> +
> +static int aic31xx_hw_params(struct snd_pcm_substream *substream,
> + struct snd_pcm_hw_params *params,
> + struct snd_soc_dai *tmp)
> +{
> + struct snd_soc_pcm_runtime *rtd = substream->private_data;
> + struct snd_soc_codec *codec = rtd->codec;
> + u8 data = 0;
> +
> + dev_dbg(codec->dev, "## %s: format %d rate %d\n",
> + __func__, params_format(params), params_rate(params));
> +
> + switch (params_width(params)) {
> + case 16:
> + break;
> + case 20:
> + data = (AIC31XX_WORD_LEN_20BITS <<
> + AIC31XX_IFACE1_DATALEN_SHIFT);
> + break;
> + case 24:
> + data = (AIC31XX_WORD_LEN_24BITS <<
> + AIC31XX_IFACE1_DATALEN_SHIFT);
> + break;
> + case 32:
> + data = (AIC31XX_WORD_LEN_32BITS <<
> + AIC31XX_IFACE1_DATALEN_SHIFT);
> + break;
> + default:
> + dev_err(codec->dev, "%s: Unsupported format %d\n",
> + __func__, params_format(params));
> + return -EINVAL;
> + }
> +
> + snd_soc_update_bits(codec, AIC31XX_IFACE1,
> + AIC31XX_IFACE1_DATALEN_MASK,
> + data);
> +
> + return aic31xx_setup_pll(codec, params);
> +}
> +
> +static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute)
> +{
> + struct snd_soc_codec *codec = codec_dai->codec;
> +
> + if (mute) {
> + snd_soc_update_bits(codec, AIC31XX_DACMUTE,
> + AIC31XX_DACMUTE_MASK,
> + AIC31XX_DACMUTE_MASK);
> + } else {
> + snd_soc_update_bits(codec, AIC31XX_DACMUTE,
> + AIC31XX_DACMUTE_MASK, 0x0);
> + }
> +
> + return 0;
> +}
> +
> +static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
> + unsigned int fmt)
> +{
> + struct snd_soc_codec *codec = codec_dai->codec;
> + u8 iface_reg1 = 0;
> + u8 iface_reg3 = 0;
> + u8 dsp_a_val = 0;
> +
> + dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt);
> +
> + /* set master/slave audio interface */
> + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
> + case SND_SOC_DAIFMT_CBM_CFM:
> + iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER;
> + break;
> + default:
> + dev_alert(codec->dev, "Invalid DAI master/slave interface\n");
> + return -EINVAL;
> + }
> +
> + /* interface format */
> + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
> + case SND_SOC_DAIFMT_I2S:
> + break;
> + case SND_SOC_DAIFMT_DSP_A:
> + dsp_a_val = 0x1;
> + case SND_SOC_DAIFMT_DSP_B:
> + /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */
> + switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
> + case SND_SOC_DAIFMT_NB_NF:
> + iface_reg3 |= AIC31XX_BCLKINV_MASK;
> + break;
> + case SND_SOC_DAIFMT_IB_NF:
> + break;
> + default:
> + return -EINVAL;
> + }
> + iface_reg1 |= (AIC31XX_DSP_MODE <<
> + AIC31XX_IFACE1_DATATYPE_SHIFT);
> + break;
> + case SND_SOC_DAIFMT_RIGHT_J:
> + iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE <<
> + AIC31XX_IFACE1_DATATYPE_SHIFT);
> + break;
> + case SND_SOC_DAIFMT_LEFT_J:
> + iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE <<
> + AIC31XX_IFACE1_DATATYPE_SHIFT);
> + break;
> + default:
> + dev_err(codec->dev, "Invalid DAI interface format\n");
> + return -EINVAL;
> + }
> +
> + snd_soc_update_bits(codec, AIC31XX_IFACE1,
> + AIC31XX_IFACE1_DATATYPE_MASK |
> + AIC31XX_IFACE1_MASTER_MASK,
> + iface_reg1);
> + snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET,
> + AIC31XX_DATA_OFFSET_MASK,
> + dsp_a_val);
> + snd_soc_update_bits(codec, AIC31XX_IFACE2,
> + AIC31XX_BCLKINV_MASK,
> + iface_reg3);
> +
> + return 0;
> +}
> +
> +static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai,
> + int clk_id, unsigned int freq, int dir)
> +{
> + struct snd_soc_codec *codec = codec_dai->codec;
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + int i;
> +
> + dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n",
> + __func__, clk_id, freq, dir);
> +
> + for (i = 0; aic31xx_divs[i].mclk != freq; i++) {
> + if (i == ARRAY_SIZE(aic31xx_divs)) {
> + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
> + __func__, freq);
> + return -EINVAL;
> + }
> + }
> +
> + /* set clock on MCLK, BCLK, or GPIO1 as PLL input */
> + snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK,
> + clk_id << AIC31XX_PLL_CLKIN_SHIFT);
> +
> + aic31xx->sysclk = freq;
> + return 0;
> +}
> +
> +static int aic31xx_regulator_event(struct notifier_block *nb,
> + unsigned long event, void *data)
> +{
> + struct aic31xx_disable_nb *disable_nb =
> + container_of(nb, struct aic31xx_disable_nb, nb);
> + struct aic31xx_priv *aic31xx = disable_nb->aic31xx;
> +
> + if (event & REGULATOR_EVENT_DISABLE) {
> + /*
> + * Put codec to reset and as at least one of the
> + * supplies was disabled. It may be to late to try
> + * soft reset, but it should not do any harm.
> + * Resetting the codec helps in minimizing leakage
> + * in case some of the regulators remains turned on.
> + */
> + if (gpio_is_valid(aic31xx->pdata.gpio_reset))
> + gpio_set_value(aic31xx->pdata.gpio_reset, 0);
> + else
> + snd_soc_write(aic31xx->codec, AIC31XX_RESET, 0x01);
> + regcache_mark_dirty(aic31xx->regmap);
> + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__);
> + }
> +
> + return 0;
> +}
> +
> +static void aic31xx_clk_on(struct snd_soc_codec *codec)
> +{
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + u8 mask = AIC31XX_PM_MASK;
> + u8 on = AIC31XX_PM_MASK;
> +
> + dev_dbg(codec->dev, "codec clock -> on (rate %d)\n",
> + aic31xx_divs[aic31xx->rate_div_line].rate);
> + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on);
> + mdelay(10);
> + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on);
> + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on);
> + if (aic31xx_divs[aic31xx->rate_div_line].nadc)
> + snd_soc_update_bits(codec, AIC31XX_NADC, mask, on);
> + if (aic31xx_divs[aic31xx->rate_div_line].madc)
> + snd_soc_update_bits(codec, AIC31XX_MADC, mask, on);
> + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on);
> + aic31xx->codec_clk_on = 1;
> +}
> +
> +static void aic31xx_clk_off(struct snd_soc_codec *codec)
> +{
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + u8 mask = AIC31XX_PM_MASK;
> + u8 off = 0;
> +
> + dev_dbg(codec->dev, "codec clock -> off\n");
> + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off);
> + snd_soc_update_bits(codec, AIC31XX_MADC, mask, off);
> + snd_soc_update_bits(codec, AIC31XX_NADC, mask, off);
> + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off);
> + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off);
> + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off);
> + aic31xx->codec_clk_on = 0;
> +}
> +
> +static int aic31xx_power_on(struct snd_soc_codec *codec)
> +{
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + int ret = 0;
> +
> + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies),
> + aic31xx->supplies);
> + if (ret)
> + return ret;
> +
> + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) {
> + gpio_set_value(aic31xx->pdata.gpio_reset, 1);
> + udelay(100);
> + }
> + regcache_cache_only(aic31xx->regmap, false);
> + ret = regcache_sync(aic31xx->regmap);
> + if (ret != 0) {
> + dev_err(codec->dev,
> + "Failed to restore cache: %d\n", ret);
> + regcache_cache_only(aic31xx->regmap, true);
> + regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies),
> + aic31xx->supplies);
> + return ret;
> + }
> + return 0;
> +}
> +
> +static int aic31xx_power_off(struct snd_soc_codec *codec)
> +{
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + int ret = 0;
> +
> + regcache_cache_only(aic31xx->regmap, true);
> + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies),
> + aic31xx->supplies);
> +
> + return ret;
> +}
> +
> +static int aic31xx_set_bias_level(struct snd_soc_codec *codec,
> + enum snd_soc_bias_level level)
> +{
> + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__,
> + codec->dapm.bias_level, level);
> +
> + switch (level) {
> + case SND_SOC_BIAS_ON:
> + break;
> + case SND_SOC_BIAS_PREPARE:
> + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
> + aic31xx_clk_on(codec);
> + break;
> + case SND_SOC_BIAS_STANDBY:
> + switch (codec->dapm.bias_level) {
> + case SND_SOC_BIAS_OFF:
> + aic31xx_power_on(codec);
> + break;
> + case SND_SOC_BIAS_PREPARE:
> + aic31xx_clk_off(codec);
> + break;
> + default:
> + BUG();
> + }
> + break;
> + case SND_SOC_BIAS_OFF:
> + aic31xx_power_off(codec);
> + break;
> + }
> + codec->dapm.bias_level = level;
> +
> + return 0;
> +}
> +
> +
extra line...
Thanks and regards,
Lokesh
> +static int aic31xx_suspend(struct snd_soc_codec *codec)
> +{
> + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
> + return 0;
> +}
> +
> +static int aic31xx_resume(struct snd_soc_codec *codec)
> +{
> + aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
> + return 0;
> +}
> +
> +static int aic31xx_codec_probe(struct snd_soc_codec *codec)
> +{
> + int ret = 0;
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + int i;
> +
> + dev_dbg(aic31xx->dev, "## %s\n", __func__);
> +
> + aic31xx = snd_soc_codec_get_drvdata(codec);
> + codec->control_data = aic31xx->regmap;
> +
> + aic31xx->codec = codec;
> +
> + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
> +
> + if (ret != 0) {
> + dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n",
> + ret);
> + return ret;
> + }
> +
> + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) {
> + aic31xx->disable_nb[i].nb.notifier_call =
> + aic31xx_regulator_event;
> + aic31xx->disable_nb[i].aic31xx = aic31xx;
> + ret = regulator_register_notifier(aic31xx->supplies[i].consumer,
> + &aic31xx->disable_nb[i].nb);
> + if (ret) {
> + dev_err(codec->dev,
> + "Failed to request regulator notifier: %d\n",
> + ret);
> + return ret;
> + }
> + }
> +
> + regcache_cache_only(aic31xx->regmap, true);
> + regcache_mark_dirty(aic31xx->regmap);
> +
> + aic31xx_add_controls(codec);
> + aic31xx_add_widgets(codec);
> +
> + return ret;
> +}
> +
> +static int aic31xx_codec_remove(struct snd_soc_codec *codec)
> +{
> + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> + int i;
> + /* power down chip */
> + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
> +
> + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
> + regulator_unregister_notifier(aic31xx->supplies[i].consumer,
> + &aic31xx->disable_nb[i].nb);
> +
> + return 0;
> +}
> +
> +static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
> + .probe = aic31xx_codec_probe,
> + .remove = aic31xx_codec_remove,
> + .suspend = aic31xx_suspend,
> + .resume = aic31xx_resume,
> + .set_bias_level = aic31xx_set_bias_level,
> + .controls = aic31xx_snd_controls,
> + .num_controls = ARRAY_SIZE(aic31xx_snd_controls),
> + .dapm_widgets = aic31xx_dapm_widgets,
> + .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets),
> + .dapm_routes = aic31xx_audio_map,
> + .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map),
> +};
> +
> +static struct snd_soc_dai_ops aic31xx_dai_ops = {
> + .hw_params = aic31xx_hw_params,
> + .set_sysclk = aic31xx_set_dai_sysclk,
> + .set_fmt = aic31xx_set_dai_fmt,
> + .digital_mute = aic31xx_dac_mute,
> +};
> +
> +static struct snd_soc_dai_driver aic31xx_dai_driver[] = {
> + {
> + .name = "tlv320aic31xx-hifi",
> + .playback = {
> + .stream_name = "Playback",
> + .channels_min = 1,
> + .channels_max = 2,
> + .rates = AIC31XX_RATES,
> + .formats = AIC31XX_FORMATS,
> + },
> + .capture = {
> + .stream_name = "Capture",
> + .channels_min = 1,
> + .channels_max = 2,
> + .rates = AIC31XX_RATES,
> + .formats = AIC31XX_FORMATS,
> + },
> + .ops = &aic31xx_dai_ops,
> + .symmetric_rates = 1,
> + }
> +};
> +
> +#if defined(CONFIG_OF)
> +static const struct of_device_id tlv320aic31xx_of_match[] = {
> + { .compatible = "ti,tlv320aic310x" },
> + { .compatible = "ti,tlv320aic311x" },
> + { .compatible = "ti,tlv320aic3100" },
> + { .compatible = "ti,tlv320aic3110" },
> + { .compatible = "ti,tlv320aic3120" },
> + { .compatible = "ti,tlv320aic3111" },
> + {},
> +};
> +MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match);
> +
> +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
> +{
> + struct device_node *np = aic31xx->dev->of_node;
> + unsigned int value = MICBIAS_2_0V;
> + int ret;
> +
> + of_property_read_u32(np, "ai31xx-micbias-vg", &value);
> + switch (value) {
> + case MICBIAS_OFF:
> + case MICBIAS_2_0V:
> + case MICBIAS_2_5V:
> + case MICBIAS_AVDDV:
> + aic31xx->pdata.micbias_vg = value;
> + break;
> + default:
> + dev_err(aic31xx->dev,
> + "Bad ai31xx-micbias-vg value %d DT\n",
> + value);
> + aic31xx->pdata.micbias_vg = MICBIAS_2_0V;
> + }
> +
> + ret = of_get_named_gpio(np, "gpio-reset", 0);
> + if (ret > 0)
> + aic31xx->pdata.gpio_reset = ret;
> +}
> +#else /* CONFIG_OF */
> +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
> +{
> +}
> +#endif /* CONFIG_OF */
> +
> +void aic31xx_device_init(struct aic31xx_priv *aic31xx)
> +{
> + int ret, i;
> +
> + dev_set_drvdata(aic31xx->dev, aic31xx);
> +
> + if (dev_get_platdata(aic31xx->dev))
> + memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev),
> + sizeof(aic31xx->pdata));
> + else if (aic31xx->dev->of_node)
> + aic31xx_pdata_from_of(aic31xx);
> +
> + if (aic31xx->pdata.gpio_reset) {
> + ret = devm_gpio_request_one(aic31xx->dev,
> + aic31xx->pdata.gpio_reset,
> + GPIOF_OUT_INIT_HIGH,
> + "aic31xx-reset-pin");
> + if (ret < 0) {
> + dev_err(aic31xx->dev, "not able to acquire gpio\n");
> + return;
> + }
> + }
> +
> + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
> + aic31xx->supplies[i].supply = aic31xx_supply_names[i];
> +
> + ret = devm_regulator_bulk_get(aic31xx->dev,
> + ARRAY_SIZE(aic31xx->supplies),
> + aic31xx->supplies);
> + if (ret != 0)
> + dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret);
> +
> +}
> +
> +static int aic31xx_i2c_probe(struct i2c_client *i2c,
> + const struct i2c_device_id *id)
> +{
> + struct aic31xx_priv *aic31xx;
> + int ret;
> + const struct regmap_config *regmap_config;
> +
> + dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__,
> + id->name, (int) id->driver_data);
> +
> + regmap_config = &aic31xx_i2c_regmap;
> +
> + aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL);
> + if (aic31xx == NULL)
> + return -ENOMEM;
> +
> + aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config);
> +
> + if (IS_ERR(aic31xx->regmap)) {
> + ret = PTR_ERR(aic31xx->regmap);
> + dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
> + ret);
> + return ret;
> + }
> + aic31xx->dev = &i2c->dev;
> +
> + aic31xx->pdata.codec_type = id->driver_data;
> +
> + aic31xx_device_init(aic31xx);
> +
> + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
> + aic31xx_dai_driver,
> + ARRAY_SIZE(aic31xx_dai_driver));
> +
> + return ret;
> +}
> +
> +static int aic31xx_i2c_remove(struct i2c_client *i2c)
> +{
> + struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev);
> +
> + kfree(aic31xx);
devm_kzalloc will take care of cleaning aic31xx during driver removal. No need to do kfree here.
> + return 0;
> +}
> +
> +static const struct i2c_device_id aic31xx_i2c_id[] = {
> + { "tlv320aic310x", AIC3100 },
> + { "tlv320aic311x", AIC3110 },
> + { "tlv320aic3100", AIC3100 },
> + { "tlv320aic3110", AIC3110 },
> + { "tlv320aic3120", AIC3120 },
> + { "tlv320aic3111", AIC3111 },
> + { }
> +};
> +MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);
> +
> +static struct i2c_driver aic31xx_i2c_driver = {
> + .driver = {
> + .name = "tlv320aic31xx-codec",
> + .owner = THIS_MODULE,
> + .of_match_table = of_match_ptr(tlv320aic31xx_of_match),
> + },
> + .probe = aic31xx_i2c_probe,
> + .remove = (aic31xx_i2c_remove),
> + .id_table = aic31xx_i2c_id,
> +};
> +
> +module_i2c_driver(aic31xx_i2c_driver);
> +
> +MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver");
> +MODULE_AUTHOR("Jyri Sarha");
> +MODULE_LICENSE("GPL");
> diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
> new file mode 100644
> index 0000000..52ed57c
> --- /dev/null
> +++ b/sound/soc/codecs/tlv320aic31xx.h
> @@ -0,0 +1,258 @@
> +/*
> + * ALSA SoC TLV320AIC31XX codec driver
> + *
> + * Copyright (C) 2013 Texas Instruments, Inc.
2014..
> + *
> + * This package is free software; you can redistribute it and/or modify
> + * it under the terms of the GNU General Public License version 2 as
> + * published by the Free Software Foundation.
> + *
> + * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR
> + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
> + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
> + *
> + */
> +#ifndef _TLV320AIC31XX_H
> +#define _TLV320AIC31XX_H
> +
> +#define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000
> +
> +#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
> + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
> +
> +
> +#define AIC31XX_STEREO_CLASS_D_BIT 0x1
> +#define AIC31XX_MINIDSP_BIT 0x2
> +
> +enum aic31xx_type {
> + AIC3100 = 0,
> + AIC3110 = AIC31XX_STEREO_CLASS_D_BIT,
> + AIC3120 = AIC31XX_MINIDSP_BIT,
> + AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT),
> +};
> +
> +struct aic31xx_pdata {
> + enum aic31xx_type codec_type;
> + unsigned int gpio_reset;
> + int micbias_vg;
> +};
> +
> +/* Page Control Register */
> +#define AIC31XX_PAGECTL 0x00
> +
> +/* Page 0 Registers */
> +/* Software reset register */
> +#define AIC31XX_RESET 0x01
> +/* OT FLAG register */
> +#define AIC31XX_OT_FLAG 0x03
> +/* Clock clock Gen muxing, Multiplexers*/
> +#define AIC31XX_CLKMUX 0x04
> +/* PLL P and R-VAL register */
> +#define AIC31XX_PLLPR 0x05
> +/* PLL J-VAL register */
> +#define AIC31XX_PLLJ 0x06
> +/* PLL D-VAL MSB register */
> +#define AIC31XX_PLLDMSB 0x07
> +/* PLL D-VAL LSB register */
> +#define AIC31XX_PLLDLSB 0x08
> +/* DAC NDAC_VAL register*/
> +#define AIC31XX_NDAC 0x0B
> +/* DAC MDAC_VAL register */
> +#define AIC31XX_MDAC 0x0C
> +/* DAC OSR setting register 1, MSB value */
> +#define AIC31XX_DOSRMSB 0x0D
> +/* DAC OSR setting register 2, LSB value */
> +#define AIC31XX_DOSRLSB 0x0E
> +#define AIC31XX_MINI_DSP_INPOL 0x10
> +/* Clock setting register 8, PLL */
> +#define AIC31XX_NADC 0x12
> +/* Clock setting register 9, PLL */
> +#define AIC31XX_MADC 0x13
> +/* ADC Oversampling (AOSR) Register */
> +#define AIC31XX_AOSR 0x14
> +/* Clock setting register 9, Multiplexers */
> +#define AIC31XX_CLKOUTMUX 0x19
> +/* Clock setting register 10, CLOCKOUT M divider value */
> +#define AIC31XX_CLKOUTMVAL 0x1A
> +/* Audio Interface Setting Register 1 */
> +#define AIC31XX_IFACE1 0x1B
> +/* Audio Data Slot Offset Programming */
> +#define AIC31XX_DATA_OFFSET 0x1C
> +/* Audio Interface Setting Register 2 */
> +#define AIC31XX_IFACE2 0x1D
> +/* Clock setting register 11, BCLK N Divider */
> +#define AIC31XX_BCLKN 0x1E
> +/* Audio Interface Setting Register 3, Secondary Audio Interface */
> +#define AIC31XX_IFACESEC1 0x1F
> +/* Audio Interface Setting Register 4 */
> +#define AIC31XX_IFACESEC2 0x20
> +/* Audio Interface Setting Register 5 */
> +#define AIC31XX_IFACESEC3 0x21
> +/* I2C Bus Condition */
> +#define AIC31XX_I2C 0x22
> +/* ADC FLAG */
> +#define AIC31XX_ADCFLAG 0x24
> +/* DAC Flag Registers */
> +#define AIC31XX_DACFLAG1 0x25
> +#define AIC31XX_DACFLAG2 0x26
> +/* Sticky Interrupt flag (overflow) */
> +#define AIC31XX_OFFLAG 0x27
> +/* Sticy DAC Interrupt flags */
> +#define AIC31XX_INTRDACFLAG 0x2C
> +/* Sticy ADC Interrupt flags */
> +#define AIC31XX_INTRADCFLAG 0x2D
> +/* DAC Interrupt flags 2 */
> +#define AIC31XX_INTRDACFLAG2 0x2E
> +/* ADC Interrupt flags 2 */
> +#define AIC31XX_INTRADCFLAG2 0x2F
> +/* INT1 interrupt control */
> +#define AIC31XX_INT1CTRL 0x30
> +/* INT2 interrupt control */
> +#define AIC31XX_INT2CTRL 0x31
> +/* GPIO1 control */
> +#define AIC31XX_GPIO1 0x33
> +
> +#define AIC31XX_DACPRB 0x3C
> +/* ADC Instruction Set Register */
> +#define AIC31XX_ADCPRB 0x3D
> +/* DAC channel setup register */
> +#define AIC31XX_DACSETUP 0x3F
> +/* DAC Mute and volume control register */
> +#define AIC31XX_DACMUTE 0x40
> +/* Left DAC channel digital volume control */
> +#define AIC31XX_LDACVOL 0x41
> +/* Right DAC channel digital volume control */
> +#define AIC31XX_RDACVOL 0x42
> +/* Headset detection */
> +#define AIC31XX_HSDETECT 0x43
> +/* ADC Digital Mic */
> +#define AIC31XX_ADCSETUP 0x51
> +/* ADC Digital Volume Control Fine Adjust */
> +#define AIC31XX_ADCFGA 0x52
> +/* ADC Digital Volume Control Coarse Adjust */
> +#define AIC31XX_ADCVOL 0x53
> +
> +
extra line here..:)
> +/* Page 1 Registers */
> +/* Headphone drivers */
> +#define AIC31XX_HPDRIVER 0x9F
> +/* Class-D Speakear Amplifier */
> +#define AIC31XX_SPKAMP 0xA0
> +/* HP Output Drivers POP Removal Settings */
> +#define AIC31XX_HPPOP 0xA1
> +/* Output Driver PGA Ramp-Down Period Control */
> +#define AIC31XX_SPPGARAMP 0xA2
> +/* DAC_L and DAC_R Output Mixer Routing */
> +#define AIC31XX_DACMIXERROUTE 0xA3
> +/* Left Analog Vol to HPL */
> +#define AIC31XX_LANALOGHPL 0xA4
> +/* Right Analog Vol to HPR */
> +#define AIC31XX_RANALOGHPR 0xA5
> +/* Left Analog Vol to SPL */
> +#define AIC31XX_LANALOGSPL 0xA6
> +/* Right Analog Vol to SPR */
> +#define AIC31XX_RANALOGSPR 0xA7
> +/* HPL Driver */
> +#define AIC31XX_HPLGAIN 0xA8
> +/* HPR Driver */
> +#define AIC31XX_HPRGAIN 0xA9
> +/* SPL Driver */
> +#define AIC31XX_SPLGAIN 0xAA
> +/* SPR Driver */
> +#define AIC31XX_SPRGAIN 0xAB
> +/* HP Driver Control */
> +#define AIC31XX_HPCONTROL 0xAC
> +/* MIC Bias Control */
> +#define AIC31XX_MICBIAS 0xAE
> +/* MIC PGA*/
> +#define AIC31XX_MICPGA 0xAF
> +/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */
> +#define AIC31XX_MICPGAPI 0xB0
> +/* ADC Input Selection for M-Terminal */
> +#define AIC31XX_MICPGAMI 0xB1
> +/* Input CM Settings */
> +#define AIC31XX_MICPGACM 0xB2
> +
> +/* Bits, masks and shifts */
> +
> +/* AIC31XX_CLKMUX */
> +#define AIC31XX_PLL_CLKIN_MASK 0x0c
> +#define AIC31XX_PLL_CLKIN_SHIFT 2
> +#define AIC31XX_PLL_CLKIN_MCLK 0
> +#define AIC31XX_CODEC_CLKIN_MASK 0x03
> +#define AIC31XX_CODEC_CLKIN_SHIFT 0
> +#define AIC31XX_CODEC_CLKIN_PLL 3
> +#define AIC31XX_CODEC_CLKIN_BCLK 1
> +
> +/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC,
> + AIC31XX_BCLKN */
> +#define AIC31XX_PLL_MASK 0x7f
> +#define AIC31XX_PM_MASK 0x80
> +
> +/* AIC31XX_IFACE1 */
> +#define AIC31XX_WORD_LEN_16BITS 0x00
> +#define AIC31XX_WORD_LEN_20BITS 0x01
> +#define AIC31XX_WORD_LEN_24BITS 0x02
> +#define AIC31XX_WORD_LEN_32BITS 0x03
> +#define AIC31XX_IFACE1_DATALEN_MASK 0x30
> +#define AIC31XX_IFACE1_DATALEN_SHIFT (4)
> +#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0
> +#define AIC31XX_IFACE1_DATATYPE_SHIFT (6)
> +#define AIC31XX_I2S_MODE 0x00
> +#define AIC31XX_DSP_MODE 0x01
> +#define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02
> +#define AIC31XX_LEFT_JUSTIFIED_MODE 0x03
> +#define AIC31XX_IFACE1_MASTER_MASK 0x0C
> +#define AIC31XX_BCLK_MASTER 0x08
> +#define AIC31XX_WCLK_MASTER 0x04
> +
> +/* AIC31XX_DATA_OFFSET */
> +#define AIC31XX_DATA_OFFSET_MASK 0xFF
> +
> +/* AIC31XX_IFACE2 */
> +#define AIC31XX_BCLKINV_MASK 0x08
> +#define AIC31XX_BDIVCLK_MASK 0x03
> +#define AIC31XX_DAC2BCLK 0x00
> +#define AIC31XX_DACMOD2BCLK 0x01
> +#define AIC31XX_ADC2BCLK 0x02
> +#define AIC31XX_ADCMOD2BCLK 0x03
> +
> +/* AIC31XX_ADCFLAG */
> +#define AIC31XX_ADCPWRSTATUS_MASK 0x40
> +
> +/* AIC31XX_DACFLAG1 */
> +#define AIC31XX_LDACPWRSTATUS_MASK 0x80
> +#define AIC31XX_RDACPWRSTATUS_MASK 0x08
> +#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20
> +#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02
> +#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10
> +#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01
> +
> +/* AIC31XX_INTRDACFLAG */
> +#define AIC31XX_HPSCDETECT_MASK 0x80
> +#define AIC31XX_BUTTONPRESS_MASK 0x20
> +#define AIC31XX_HSPLUG_MASK 0x10
> +#define AIC31XX_LDRCTHRES_MASK 0x08
> +#define AIC31XX_RDRCTHRES_MASK 0x04
> +#define AIC31XX_DACSINT_MASK 0x02
> +#define AIC31XX_DACAINT_MASK 0x01
> +
> +/* AIC31XX_INT1CTRL */
> +#define AIC31XX_HSPLUGDET_MASK 0x80
> +#define AIC31XX_BUTTONPRESSDET_MASK 0x40
> +#define AIC31XX_DRCTHRES_MASK 0x20
> +#define AIC31XX_AGCNOISE_MASK 0x10
> +#define AIC31XX_OC_MASK 0x08
> +#define AIC31XX_ENGINE_MASK 0x04
> +
> +/* AIC31XX_DACSETUP */
> +#define AIC31XX_SOFTSTEP_MASK 0x03
> +
> +/* AIC31XX_DACMUTE */
> +#define AIC31XX_DACMUTE_MASK 0x0C
> +
> +/* AIC31XX_MICBIAS */
> +#define AIC31XX_MICBIAS_MASK 0x03
> +#define AIC31XX_MICBIAS_SHIFT 0
> +
> +#endif /* _TLV320AIC31XX_H */
>
More information about the Alsa-devel
mailing list